Vehicular microphone system and method for post processing optimization of a microphone signal

ABSTRACT

A vehicular microphone system ( 200 ) for post processing optimization of a microphone signal includes a first transducer ( 201 ) and second transducer ( 203 ) separated by a predetermined distance within an automotive mirror. A first high pass filter network ( 205 ) is connected to the first transducer ( 201 ) while a second high pass filter network ( 207 ) connected to the second transducer ( 203 ). A low frequency shelving filter ( 209 ) is used for receiving the output from the second high pass filter ( 207 ). A first all pass filter ( 211 ) is connected to the low frequency shelving filter ( 209 ) and a second all pass filter ( 213 ) is used in connection with the first all pass filter ( 211 ) for tailoring audio characteristics. A summing amplifier ( 215 ) is used for summing the output of the first high pass filter network ( 201, 205 ) and the second all pass filter network ( 203, 207, 209, 211, 213 ) such that the first transducer ( 201 ) and second transducer ( 203 ) operate with improved directivity resulting in enhanced signal-to-noise performance in a substantially noisy vehicular environment.

CROSS REFERENCE TO RELATED APPLICATIONS

This Application claims priority of U.S. Provisional Application No.61/122,548, filed on Dec. 14, 2009.

FIELD OF THE INVENTION

The present invention pertains to microphones and more particularly to amicrophone arrangement associated with a vehicle accessory.

BACKGROUND

Microphones used in automotive electronic applications, such as cellphones, navigational systems, and vehicular control, are well-known inthe art. An automotive vehicle presents challenges to the use of amicrophone in view of the numerous sources of noise that can interferewith vocalized speech inside the vehicle. These challenges can beparticularly difficult when adapting a microphone solution for use inthe vehicular rearview mirror assembly. In addition to the difficultiesof rejecting noise within the vehicle, disturbances to the sound fieldcaused by the rearview mirror must also be addressed.

The prior art includes systems that use microphones positioned intandem, i.e., a first microphone positioned in front of a secondmicrophone. This type of system works to produce a difference signal forcanceling noise by subtracting the signals and using a delay to accountfor the distance between the microphones. However, the rearview mirrordisturbs the sound field between the two microphones, which results inpoor subtraction over much of the frequency range of interest.Additionally, this front and back microphone configuration requires therearview mirror to include a deeper housing for supporting the rearwardmicrophone, which is often an undesired design feature in view ofstyling, weight, vibration sensitivity, and molding required in themanufacturing process.

Other prior art systems use microphones positioned in parallel that usedigital processing or simple delay networks to improve operation. Theuse of digital processing introduces delay and variation over time thatdisrupts systems designed for a single microphone. Therefore, this typeof simple delay based processing does not yield the desired performance.Additionally, many of the microphone systems currently in use weredeveloped under the assumption that the microphone would be used inconnection with a handheld cellular phone. In handheld applications, thevery close proximity of the user's mouth to the microphone assures avery high speech-to-noise content for most situations. These systems donot function correctly with microphones used at a distance because audioreceived at increased distances does not exhibit the same frequencycharacteristics.

Microphones distant from an audio source that are used in a hands-freecar system, unlike a very close use situation, will often have a verysignificant noise content, and manifest a wider dynamic range. A “closeuse” situation or microphone may be defined as one positioned within 20cm of the audio source such as a user's mouth. The dynamic range isincreased because of the broader range of possible speech signal levelsand relative noise content. In a distant use situation, if a widerdynamic range speech signal is processed via the phone system,especially phones employing code division multiple access (CDMA), muchof the desired speech can be lost because the processing system isunable to correctly determine that speech is present. Thus, the phonesystem functions as if a voice signal is comprised of only noise.

The current state of the art seeks to lower the noise content whileretaining the speech in its unaltered state. This process does notrestore the nature of the speech signal to that of a close usemicrophone as found in a typical handset and as a result does not yielda signal able to pass through the cell phone's CODEC. As a result, therewill be many frequency bands or occurrences where the speech content,though significant, is not great enough to overcome the residual noiseto the extent so as to avoid being interpreted as noise. Thus, in latterprocessing stages, these frequency bands or occurrences will be removedbecause they appear to be only unwanted noise. Even though the speechcontent is significant, it is not of a great enough magnitude toovercome the noise in certain frequency bands or at certain times.

Moreover, most cars driven above 50 mph, on rough roads, will have lessthan acceptable speech quality through the cellular phone when using aBluetooth connected hands-free microphone system because of limiteddynamic range. These problems are magnified where the cellular phone isa CDMA type because the CODECs employed in these phones are lesstolerant of a wide dynamic range signal. The CODEC system, in attemptingto limit band width, stops correctly transmitting the speech signalbecause it interprets it as being unwanted noise. In some cases, thespeech components in the cellular call can be totally lost. The loss canbe such that the user may feel they have suffered a classic cellularphone drop out when, in fact, the call is still in progress andconnected. Since there are literally millions of cellular phones withCODECs implementing a bandwidth/noise reduction algorithm that willtruncate speech as described, the only hope for a solution is a processthat will result in a signal whose speech content will be passed evenwhen noise is also present.

There are three aspects that can address incorrect CODEC operation,processing of the microphone signal to emulate the signal from a closeused microphone, reduction of the noise proportion in a way that doesnot otherwise harm system operation, and an elevation of all significantcontent speech frequency bands to a magnitude well above all otherbands. In every case the threshold where an action is invoked is can bevariable based on a number of factors including the average noise level,peak noise levels, least noise level, average speech levels, leastspeech level and peak speech levels. The intent being to process thespeech carrying signal to minimize the impact of any noise present. Thisprocessing ideally needs to be adjusted to align with the conditionspresent at that time.

BRIEF DESCRIPTION OF THE FIGURES

The accompanying figures, where like reference numerals refer toidentical or functionally similar elements throughout the separate viewsand which together with the detailed description below are incorporatedin and form part of the specification, serve to further illustratevarious embodiments and to explain various principles and advantages allin accordance with the present invention. These figures address oneaspect of the three groups involved, the use of two transducers toreduce the noise content in the speech signal. The other two conceptsare defined via textual descriptions and can be used in conjunction withthe dual microphone processing or exclusively in various combinations.

FIG. 1 is a perspective view of a rearview mirror assembly using avehicular microphone system in accordance with an embodiment of theinvention.

FIG. 2 is block diagram of a two-microphone array in accordance with anembodiment of the invention.

FIG. 3 is a block diagram illustrating microphone positioning of themicrophone array in accordance with an embodiment of the invention.

FIG. 4 is a block diagram illustrating flowchart diagram illustrating aprocess for post processing optimization of a microphone signalcontaining residual noise in accordance with an embodiment of theinvention.

Skilled artisans will appreciate that elements in the figures areillustrated for simplicity and clarity and have not necessarily beendrawn to scale. For example, the dimensions of some of the elements inthe figures may be exaggerated relative to other elements to help toimprove understanding of embodiments of the present invention.

These and other features, advantages, and objects of the presentinvention will be further understood and appreciated by those skilled inthe art by reference to the following specification, claims, andappended drawings.

DETAILED DESCRIPTION

Before describing in detail embodiments that are in accordance with thepresent invention, it should be observed that the embodiments resideprimarily in combinations of method steps and apparatus componentsrelated to post processing optimization of a microphone signalcontaining residual noise. Accordingly, the apparatus components andmethod steps have been represented where appropriate by conventionalsymbols in the drawings, showing only those specific details that arepertinent to understanding the embodiments of the present invention soas not to obscure the disclosure with details that will be readilyapparent to those of ordinary skill in the art having the benefit of thedescription herein.

In this document, relational terms such as first and second, top andbottom, and the like may be used solely to distinguish one entity oraction from another entity or action without necessarily requiring orimplying any actual such relationship or order between such entities oractions. The terms “comprises,” “comprising,” or any other variationthereof, are intended to cover a non-exclusive inclusion, such that aprocess, method, article, or apparatus that comprises a list of elementsdoes not include only those elements but may include other elements notexpressly listed or inherent to such process, method, article, orapparatus. An element proceeded by “comprises . . . a” does not, withoutmore constraints, preclude the existence of additional identicalelements in the process, method, article, or apparatus that comprisesthe element.

It will be appreciated that embodiments of the invention describedherein may be comprised of one or more conventional processors andunique stored program instructions that control the one or moreprocessors to implement, in conjunction with certain non-processorcircuits, some, most, or all of the functions of post processingoptimization of a microphone signal containing residual noise. Thenon-processor circuits may include, but are not limited to, signalamplifiers, clock circuits, power source circuits, compressors,expandors, noise gates, and user input devices. Those skilled in the artwill recognize that compressors, expandors and/or noise gates can beimplemented in several ways, including the use of voltage controlledamplifiers. As such, these functions may be interpreted as steps of amethod to perform post processing optimization of a microphone signalcontaining residual noise and/or excessive dynamic range. Alternatively,some or all functions could be implemented by a state machine that hasno stored program instructions, or in one or more application specificintegrated circuits (ASICs), in which each function or some combinationsof certain of the functions are implemented as custom logic. Of course,a combination of the two approaches could be used. Thus, methods andmeans for these functions have been described herein. Further, it isexpected that one of ordinary skill, notwithstanding possiblysignificant effort and many design choices motivated by, for example,available time, current technology, and economic considerations, whenguided by the concepts and principles disclosed herein will be readilycapable of generating such software instructions and programs and ICswith minimal experimentation.

FIG. 1 is a perspective view of a rearview mirror assembly using avehicular microphone system in accordance with an embodiment of theinvention. The rearview microphone assembly 100 includes a housing 101that is supported within the interior of the vehicle. A mirror 103 isused by the driver to view objects from his rear. A first transducer 105and second transducer 107 are positioned within the housing 101 and areused to capture voice from inside a vehicle.

FIG. 2 is block diagram of a two-microphone array 200 using a first andsecond channel in accordance with an embodiment of the invention. Afirst transducer 201 and second transducer 203 are used in the array andseparated by some predetermined distance. The first transducer 201 andsecond transducer 203 are audio microphones or the like. The firsttransducer 201 is connected to a first high pass filtering network 205,while the second transducer 203 is connected to a second high passfilter network 207. Both the first high pass filtering network 205 andsecond high pass filtering network 207 include both filtering andamplification circuitry for tailoring the audio signal in apredetermined fashion. Both filter and amplifier circuits need not beidentical, however, and may, in fact, filter and/or amplify signals fromtheir respective transducers in a unique manner with differing filtercut-off frequencies and characteristics.

In the second channel, the high pass filter network 207 is fed to a lowfrequency shelving filter that works to compensate for the unequalfrequency response from the high pass filter 207. The shelving filterallows one or more parameters of the audio signal from the high passfilter network 207 to be adjusted for determining the overall shape ofthe filter's transfer function. The shelving filter ultimately improvesthe fidelity of sound, to emphasize certain voice characteristics and toremove undesired noises in the vehicle. The output of the low frequencyshelving filter 209 is then directed to a first all pass filter 211. Theall pass filter 211 is an electronic filter that passes all frequenciesequally, but changes the phase relationship between various frequencies.It does this by varying its propagation delay with frequency. Generally,the all pass filter 211 can be described by the frequency at which thephase shift crosses 90 degrees, such as in this case 180 degrees to 0degrees. The all pass filter 211 is used to compensate for the undesiredphase shifts that arise in the system from second high pass filternetwork 207 and the low frequency shelving filter 209. The output of thefirst all-pass filter is then directed to a second all pass filter 213that operates where the phase shift crosses 90 degree between 0 degreesand 180 degrees. The output of the first high pass filtering network 205in the first channel and the second all pass filter 213 in the secondchannel are input to a summing amplifier and low pass filter 215 whichacts to increase the magnitude of the summed signal to further reducehigh frequency components. The output of the summing amplifier and lowpass filter 215 is directed to an optional filtering network for furthertailoring frequency response.

Hence, the improved vehicular microphone array system 200 utilizes bothacoustic and electrical delay and summing of multiple signals to achievebetter noise reduction. It will be recognized that one of the keyconditions in using microphone arrays is that the preferred signalsreceived by the microphone transducers must be identical in order toutilize linear subtraction for noise reduction. This requirementpresents a real challenge in an automotive vehicle as the small enclosedspace creates numerous reflections and disturbances of the sound field.In particular, when microphones are mounted on the rearview mirror in avehicle, the shape and position of the mirror with respect to surfaces,such as the windshield, headliner, and such, results in large acousticpressure differences across the various surfaces of the mirror. It hasbeen found that the sound pressure difference for a sound originatingtoward the front of the mirror, between the front and back of a typicalrearview mirror, can be in excess of 10 dB at some frequencies, whichmakes array processing all but useless at those frequencies. There arealso large phase differences as well caused by the shape and position ofthe mirror in the vehicle. Thus the design has to address these acousticrealities for this approach to operate correctly.

In order for the rearview mirror to perform its primary function, themirror includes a large flat reflective glass surface which has a largeaspect ratio, such that it is wide, about one third as tall as thewidth, and only as deep as required to conceal any accessories such asmap lights or other electronics. This specific shape causes any soundfield projected at the mirror to develop a large pressure increaseacross the front surface for sounds having at least ¼ wavelength of thedimensions of the mirror reflective surface. When a microphone ismounted on the surfaces behind the reflective surface, the soundpressure typically drops the farther away it is located from the frontof the mirror. This effect is beneficial for some frequencies, and somemanufacturers have proposed mounting the microphone on or close to thefront surface of the mirror. This presents difficult challenges withrespect to manufacturing, especially with a directional transducer, andis often not a preferred location for a mirror mounted microphone. Anembodiment of the invention addresses this shortcoming by achieving verygood performance utilizing microphones mounted back from the glasssurface.

FIG. 3 is a block diagram illustrating a multiple microphone array 300in accordance with an embodiment of the invention. In this example, thefirst microphone 301 and the second microphone 303 are positionedapproximately 46 mm apart with an offset of approximately 15 degreesfrom the driver 305. Those skilled in the art will recognize that whenmultiple microphones are mounted on a mirror, the microphones must beinstalled such that they are the same distance from the front reflectivesurface to have a nearly equal on-axis frequency response. Also, becauseof the aspect ratio of the mirror, the microphones should be positionedin relatively close proximity to have a nearly equal off-axis frequencyresponse. This creates a tradeoff between a greater spacing's improvedlower frequency performance and avoidance of secondary lobes in thehigher frequencies caused by too great a spacing.

Therefore, small spacing distances lend to good high frequencydirectionality, while large spacing distances are required for good lowfrequency directionality. The spacing for the microphones is, therefore,in the range of less than 102 mm, with 50 mm being a nearly optimalvalue, for an automotive rearview mirror mounted microphone array. Thoseskilled in the art will recognize that when mounted in a typical carinterior, the reflective surfaces cause an usually large drop in theon-axis frequency response of the microphone in the 800 to 1500 Hz rangeon the side of the mirror away from the driver. This characteristic addsan additional reason for rejecting the use of a wide spacing, such as102 mm, since it is not desirable for a high microphone performancedesign.

One problem involved in reducing vehicular noise using digital signalprocessing (DSP) is that the frequency bands identified as havingsignificant speech content are often below the maximum allowedmagnitude. When the reconstructed signal is subsequently processed by acellular phone or telephone network, the voice audio signal is onceagain broken down into frequency bands that typically are fewer than wasdone in the initial DSP process. Any speech in bands that is of too lowa magnitude will be filtered and/or removed by this process. The resultis speech components that are no longer usable or, in more seriouscases, are totally missing.

FIG. 4 is a block diagram illustrating flowchart diagram illustrating aprocess for post processing optimization of a microphone signalcontaining residual noise 400 in accordance with an embodiment of theinvention. Initially, the process begins where the noise level isestimated 401. Most noise processing systems are capable of determiningand/or using a noise estimate; therefore, the estimate may also bedetermined from the existing processing 403 that proceeds the noiselevel estimation 401. The estimated noise level is then used todetermine if exceptional processing is required, i.e., the noise levelis at a substantial level in order to warrant the use of additionalprocessing 405. Since this processing will alter the normal audiocharacteristics, additional processing of the audio typically will notbe performed unless the noise level exceeds some predeterminedthreshold.

If no additional processing is required, the audio in the frequencydomain will be applied to an inverse FFT process 411 in order toreproduce the most natural sound of the user's voice. In the eventexceptional digital signal processing (DSP) is required to avoiddownstream errors, then the process operates to increase every bandabove the estimated noise by an amount that preserves the relativemagnitudes 407. This process prevents the CODEC from eliminating anyfrequency band with significant speech content. This process can beperformed by establishing a constant for a percentage of the differencebetween the actual band magnitude and the noise estimate. For example,if a 10 percent factor is desired and there is a 100 unit differencebetween the maximum tolerable value and the desired frequency band, thisdifference would be reduced to a 10 unit difference. Thus, the smallerthe percentage of retention results the greater reduction in dynamicrange. The lower energy bands being elevated to nearly the samemagnitude as the most intense.

Those skilled in the art will further recognize that a key aspect ofthis process is to preserve the relative signal magnitudes to assureacceptable sounding speech. It is also possible the compression factorwill vary with the noise estimate being a smaller percentage as thenoise rises and greater compression is required. Subsequently, after themagnitudes of the frequency bands having the noise are increased, thefrequency bands whose magnitudes are below this level are then decreasedbelow the desired noise estimate 409. Hence, the magnitude of anyfrequency band below the estimate is reduced to a very low,insignificant, value 407. This has the effect of increasing the apparentsignal-to-noise ratio. The noise content reference is reduced in thedownstream stages when the processed audio is subjected to an inverseFFT process 411 for conversion to the time domain.

Therefore, the process described in FIG. 4 operates to raise themagnitude of all speech bands near their maximum value. When processeddownstream by a subsequent cellular telephone or a telephone network,all of the speech bands will be passed as they are above somepredetermined threshold magnitude. This increase in magnitude of thespeech bands does not result in distorted speech, nor does clipping thebands at the maximum level result in any gross distortion products.Raising the magnitude of the weaker audio bands, i.e., those less inmagnitude, is generally interpreted by those listening to the call as aperson raising his/her voice in high noise environment, an acceptablechange in most cases.

In operation, the speech process in accordance with the presentinvention works to elevate specific frequency bands containing“significant” speech. This works to make all of the resulting audiothough a cellular handset be interpreted by a listener as only normalspeech irrespective of this type of unique signal processing. Inpractice, the relative relationship between speech content frequencybands should be maintained, but merely compressed toward the maximumvalue which makes the resulting processed speech sound like normalaudio. Ideally, the processing, as described herein, will only beperformed in situations where noise is present and at a degree ofprocessing that reflects the current noise level. No processing isperformed in quiet conditions but instead only near frequency bandclipping levels under conditions where the highest noise is present.

The process as described herein operates to pass noise that is elevatedin magnitude by some predetermined level since it is preferred to pass aminimal amount of noise rather than to lose important speech content. Anundesired “noise pumping” can be reduced by adding white noise i.e.noise having a wide range frequency spectrum. Finally, a determinationof band gain can be performed in several ways where a nonlinear gain isused to raise the weaker frequency bands substantially close to theirmaximum value while avoiding gross clipping of the more powerfulfrequency bands which will retain the relative magnitudes. Under theseconditions, the voice bands having the greatest magnitude will remainthe highest post audio processing, while the frequency bands of thelowest or weakest magnitude will still be weaker although the differencewill be reduced. Those skilled in art will recognize that this processoperates in a unique manner to mimic some processing of a complexcompressor/expander.

In yet another embodiment, the threshold can be varied for expansionand/or compression (compandor). Although a conventional expandor orcompressor has fixed thresholds, these bandwidths can be made adaptivelydynamic for optimal performance. In effect, the magnitude of the signalthat triggers the event will vary with its use condition. In quietconditions, where both the noise and speech are relatively low, thesignal magnitude required to cause expansion will be lower than insubstantially high noise and higher average speech level conditions.Accordingly, this solution works to pass all speech while stillrejecting a significant amount of the noise. In a fixed thresholdsystem, the lowest speech level condition determines the threshold toassure all speech is passed. Ideally this threshold boundary willincrease as both the speech amplitude and noise increase. The preferredor ideal level should be just at the boundary to assure passage of allprobable speech levels.

Hence, an embodiment of the invention is directed to a multiplemicrophone assembly with analog processing circuitry for providinggreater directivity resulting in improved signal-to-noise performance inview of a noisy environment present in automotive vehicles. As describedherein, microphones mounted within the vehicle are placed parallel toone another, so as to enhance performance when mounted on a vehicleaccessory such as a mirror. An embodiment of the invention works with aprocess to address the loss of speech due to residual noise indownstream audio processes, such as those in the cellular telephonehandset and/or network systems. In addition, this approach reduces theimpact of dynamic range limitations imposed by a Bluetooth remotemicrophone link while also addressing the impact on cellular telephonesystems designed for typical headset speech signals by emulating thesecharacteristics.

Aspects of this invention are based on at least three observations,namely, the speech frequency bands in the frequency domain can beclipped and the resulting speech remains intelligible, persons withoccupations requiring public speaking in a high noise environment findthat speaking using a vocal cord frequency that will offer the greatestnumber of frequency bands at a elevated level, and the characteristicsof a cellular handset's microphone are required for proper function bysome cellular networks. The persons speaking in high noise environmentssound as if they are talking loud but otherwise sound normal. Takentogether, the new system and methods as described herein apply gain tothe speech bands or speech times raising them in amplitude near to themaximum level and reducing significantly bands or times when the speechlevels are below this established value. This is in contrast to typicalsignal processing which leaves the speech content in processed signalsas it is captured in high noise situations. These actions assure thatthe appropriate and/or most important voice components will not beremoved by downstream audio signal processing. Although, the resultantspeech does not sound completely normal, but instead sounds more like aperson speaking loudly, this resulting sonic characteristic isacceptable to most cellular telephone users. Further, if the relativemagnitude of the speech bands is maintained, the speech sounds morenatural. Although there can be a degree of noise pumping using thisprocess, such as a variation in noise with the speech, this result ispreferable to the loss of critical speech frequency components. Anynoise pumping can also be reduced using the methods are describedherein.

In the foregoing specification, specific embodiments of the presentinvention have been described. However, one of ordinary skill in the artappreciates that various modifications and changes can be made withoutdeparting from the scope of the present invention as set forth in theclaims below. Accordingly, the specification and figures are to beregarded in an illustrative rather than a restrictive sense, and allsuch modifications are intended to be included within the scope ofpresent invention. The benefits, advantages, solutions to problems, andany element(s) that may cause any benefit, advantage, or solution tooccur or become more pronounced are not to be construed as a critical,required, or essential features or elements of any or all the claims.The invention is defined solely by the appended claims including anyamendments made during the pendency of this application and allequivalents of those claims as issued.

We claim:
 1. A method for post processing optimization of a microphonesignal used in a vehicle mirror comprising the steps of: generating asignal containing both voice and noise from at least one microphonelocated in a vehicle; estimating a level of the noise present within thesignal; determining if the level of the noise is above a predeterminedthreshold; modifying the signal to increase the magnitude of both thevoice and the noise for those frequency bands in the signal having alevel of noise above the predetermined threshold; modifying the signalto decrease the magnitude of both the voice and noise for thosefrequency bands in the signal having a level of noise below thepredetermined threshold; and submitting the modified signal to a postprocessing network.
 2. A method for post processing optimization as inclaim 1, further comprising the step of: converting the generated signalinto the frequency domain before estimating the level of noise.
 3. Amethod for post processing optimization as in claim 1, wherein theposting processing network is an inverse fast Fourier transform (FFT)for converting the modified signal into the time domain.
 4. A vehicularmicrophone system for post processing optimization of a microphonesystem comprising: at least one microphone configured to receive a voiceand a noise from inside the vehicle and generate a signal; a digitalsignal processor configured to: determine a noise level present withinthe signal; determine if the noise level is above or below apredetermined threshold; modify the signal to increase a magnitude ofboth the voice and noise for those frequency bands in the signal havinga level of noise above the predetermined threshold; modify the signal todecrease the magnitude of both the voice and noise for those frequencybands in the signal having a level of noise below the predeterminedthreshold: and submit the modified signal to a post processing network.5. A vehicle microphone system as in claim 4, wherein the at least onemicrophone comprises a first transducer and a second transducerseparated by a predetermined distance.
 6. A vehicle microphone system asin claim 5 further comprising a first and second high pass filternetwork, wherein the first high pass filter network is operablyconnected to the first transducer and the second high pass filternetwork is operably connected to the second transducer.
 7. A vehiclemicrophone system as in claim 6 further comprising a low frequencyshelving filter configured to receive the output from the second highpass filter.
 8. A vehicle microphone system as in claim 7 furthercomprising a first all pass filter operably connected to the lowfrequency shelving filter.
 9. A vehicle microphone system as in claim 8further comprising a second all pass filter operably connected to thefirst all pass filter.
 10. A vehicle microphone system as in claim 9further comprising a summing amplifier configured to receive the outputof the first high pass filter network and the second all pass filter.11. A vehicle microphone system as in claim 10, wherein the firsttransducer and second transducer are configured to operate with improveddirectivity resulting in enhanced signal-to-noise performance in asubstantially noisy vehicular environment.